The present invention relates to a method of decoding digital audio data.
It is conventional that with the digital radio transmission method referred to as DAB (Digital Audio Broadcasting), source decoding includes error detection and correction, dequantization and filtering of data. In a related art channel coding, error detecting and correcting codes are used, whereas a checksum (cyclic redundancy check=CRC) is used in decoding the digital audio data, and when an error is detected, the data containing the error is replaced by equivalent preceding data.
U.S. patent application Ser. No. 5,450,081 describes error concealment in the time range on the analog audio signal. European Patent Application No. 0 718 982 describes error concealment in general with equalization of audio data by frames. Patent Abstracts of Japan JP 5328290 describes error concealment by interpolation when a count is reached or exceeded. DAB and various error concealment techniques are described by D. Wiese in xe2x80x9cOptimization of Error Detection and Concealment for ISO/MPEG/AUDIO CODECs Layer I and II,xe2x80x9d 93rd AES Convention, no. 3368, 1992, pages 1-18. For example, replacement of corrupted scale factors by scale factors previously received correctly is mentioned. Another method describes filtering of high frequencies, because the interference there is localized to an especially great extent.
The method according to the present invention for decoding digital audio data may provide the advantage that spectral shaping of audio signals is performed during dequantization as a function of an error number determined by using the checksum. Errors that occur are compensated to advantage in this manner by using the error number to estimate how the audio spectrum must be altered to minimize the effects of these errors. Errors are concealed in this manner.
The method according to the present invention may have a low additional cost and may be implemented in any audio decoder. In particular the fact that the errors are concealed individually results in a gradual loss of quality which is not otherwise possible with digital data. This is pleasant for a listener, although a loss of quality would nevertheless be noticed.
It may be advantageous that the values are either loaded from a memory and/or calculated by a processor. This makes use of knowledge with which the stored equalizer values were originally determined, and on the other hand, the equalizer values may be adapted to the respective situation by a calculation, thus achieving an adaptive performance characteristic. The error correction is thus adapted optimally, so that the user of a radio receiver using the method according to the present invention will not notice a sudden decline in quality of the audio signals.
In addition, it may be advantageous that the measure of the quality of the digital audio data is compared with threshold values. This makes it possible to set corresponding equalizer values as a function of whether or not this measure is above preselected threshold values. This permits a simple adaptation to the respective error situation. The method according to the present invention is not used when this measure indicates a very low error number or freedom from errors, because no error correction is necessary. If this measure indicates an error number in excess of the largest threshold value, i.e., the error correction no longer offers a remedy, then a muting is activated. Thus an optimized error correction is offered to the user as a function of the error number.